This is an archived snapshot of W3C's public bugzilla bug tracker, decommissioned in April 2019. Please see the home page for more details.
This archived bug list includes all the bugs for the product or component specified, including bugs that have been resolved.
ID | Product | Comp | Assignee | Status | Resolution | Summary | Opened | Changed |
---|---|---|---|---|---|---|---|---|
15205 | WebRTC W | WebRTC A | fluffy | RESO | FIXE | Add API for sending and playing DTMF tones | 2011-12-15 | 2012-03-19 |
15206 | WebRTC W | WebRTC A | juberti | RESO | FIXE | Add API for sending and receiving p2p application data | 2011-12-15 | 2012-03-19 |
15593 | WebRTC W | WebRTC A | adam.bergkvist | RESO | FIXE | Update PeerConnection functionality associated to adding/removing MediaStreams | 2012-01-17 | 2012-05-21 |
15729 | WebRTC W | WebRTC A | public-webrtc | RESO | INVA | The editors draft doesn't specify what should happen if a null stream is added to a PeerConnection | 2012-01-26 | 2012-01-27 |
15747 | WebRTC W | WebRTC A | fluffy | RESO | FIXE | Echo cancellation API | 2012-01-27 | 2012-05-18 |
15861 | WebRTC W | WebRTC A | public-webrtc | RESO | LATE | API for JS interaction with congestion control | 2012-02-03 | 2014-10-31 |
16273 | WebRTC W | WebRTC A | public-webrtc | RESO | FIXE | MediaStreamRecorder interface should be removed until redesigned | 2012-03-08 | 2012-05-18 |
17109 | WebRTC W | WebRTC A | public-webrtc | RESO | FIXE | TURN server API changes | 2012-05-18 | 2014-04-22 |
17243 | WebRTC W | WebRTC A | public-webrtc | RESO | INVA | suggestion on "select camera function for getUserMedia()" | 2012-05-30 | 2013-01-29 |
17249 | WebRTC W | WebRTC A | public-webrtc | RESO | FIXE | SdpType usage must be chosen | 2012-05-30 | 2012-08-13 |
17287 | WebRTC W | WebRTC A | public-webrtc | RESO | FIXE | PeerConnectionErrorCallback argument | 2012-06-01 | 2014-10-31 |
17596 | WebRTC W | WebRTC A | public-webrtc | RESO | WONT | Can we have an method fast peerconnection reconstruction? | 2012-06-26 | 2012-08-03 |
18335 | WebRTC W | WebRTC A | public-webrtc | RESO | FIXE | PeerConnection should inherit EventTarget | 2012-07-19 | 2012-08-13 |
18443 | WebRTC W | WebRTC A | public-webrtc | RESO | INVA | getUserMedia() Test Error in Chrome | 2012-07-31 | 2012-08-14 |
18485 | WebRTC W | WebRTC A | public-webrtc | RESO | FIXE | Change DTMF API to be on PeerConnection | 2012-08-03 | 2013-06-13 |
18486 | WebRTC W | WebRTC A | adam.bergkvist | RESO | FIXE | Let RTCSessionDescription take a Dictionary parameter | 2012-08-04 | 2012-09-13 |
18516 | WebRTC W | WebRTC A | public-webrtc | RESO | FIXE | MediaStream should extend EventTarget, not mix it in | 2012-08-10 | 2012-10-26 |
18869 | WebRTC W | WebRTC A | adam.bergkvist | RESO | FIXE | DataChannel should inherit EventTarget | 2012-09-13 | 2012-09-13 |
18889 | WebRTC W | WebRTC A | public-webrtc | RESO | INVA | RequestWWW | 2012-09-15 | 2012-09-15 |
19580 | WebRTC W | WebRTC A | public-webrtc | RESO | FIXE | Callbacks need to be called asynchronously | 2012-10-17 | 2014-10-31 |
19587 | WebRTC W | WebRTC A | public-webrtc | RESO | FIXE | change "DataChannelEvent" to "RTCDataChannelEvent" | 2012-10-18 | 2012-10-25 |
19588 | WebRTC W | WebRTC A | public-webrtc | RESO | WONT | change "MediaStreamEvent" to "RTCMediaStreamEvent" | 2012-10-18 | 2012-10-26 |
19592 | WebRTC W | WebRTC A | public-webrtc | RESO | FIXE | move section 6 IANA registration to end of specification | 2012-10-18 | 2012-10-26 |
19593 | WebRTC W | WebRTC A | fluffy | RESO | FIXE | create reference for mediaconstraints in createOffer() method | 2012-10-18 | 2014-10-28 |
19717 | WebRTC W | WebRTC A | public-webrtc | RESO | FIXE | Remove "number hints" from enum references (e.g. "closed (3)"). | 2012-10-26 | 2012-10-26 |
19729 | WebRTC W | WebRTC A | dom | RESO | LATE | missing a reference for XMLHttpRequest in 4.1 Introduction | 2012-10-27 | 2014-12-15 |
19730 | WebRTC W | WebRTC A | public-webrtc | RESO | FIXE | suggest to remove issue 1 from 4.3.1 | 2012-10-27 | 2014-10-28 |
19731 | WebRTC W | WebRTC A | public-webrtc | RESO | FIXE | wrong transition description between "checking" and "connected" in 4.4.3 | 2012-10-27 | 2014-10-28 |
19857 | WebRTC W | WebRTC A | public-webrtc | RESO | FIXE | onnegotationneeded syntax error | 2012-11-05 | 2012-11-05 |
20794 | WebRTC W | WebRTC A | public-webrtc | RESO | INVA | i want to connect p2p with html5 | 2013-01-28 | 2013-06-13 |
20796 | WebRTC W | WebRTC A | public-webrtc | RESO | WONT | javascript identifier for each available camera | 2013-01-28 | 2013-10-10 |
20806 | WebRTC W | WebRTC A | dom | RESO | LATE | Section 15 (Security Considerations) is empty | 2013-01-29 | 2014-12-15 |
20807 | WebRTC W | WebRTC A | public-webrtc | RESO | FIXE | End-to-end stream configuration unspecified | 2013-01-29 | 2014-10-28 |
20808 | WebRTC W | WebRTC A | public-webrtc | RESO | FIXE | "Trickle ICE" unspecified | 2013-01-29 | 2014-10-28 |
20809 | WebRTC W | WebRTC A | adam.bergkvist | RESO | LATE | Stream rejection not possible | 2013-01-29 | 2014-12-15 |
20810 | WebRTC W | WebRTC A | public-webrtc | RESO | WONT | SDP inadequately defined | 2013-01-29 | 2014-10-28 |
20811 | WebRTC W | WebRTC A | public-webrtc | RESO | FIXE | RTP usage not defined | 2013-01-29 | 2014-10-31 |
20812 | WebRTC W | WebRTC A | public-webrtc | RESO | FIXE | Media protocol not specified | 2013-01-29 | 2014-10-28 |
20813 | WebRTC W | WebRTC A | public-webrtc | RESO | FIXE | Protocol security not defined | 2013-01-29 | 2014-10-28 |
20814 | WebRTC W | WebRTC A | public-webrtc | RESO | FIXE | Data channel protocol not specified | 2013-01-29 | 2014-04-29 |
20815 | WebRTC W | WebRTC A | public-webrtc | RESO | FIXE | Stream multiplexing | 2013-01-29 | 2014-10-28 |
20816 | WebRTC W | WebRTC A | juberti | RESO | LATE | "Hold" unspecified | 2013-01-29 | 2014-12-15 |
20817 | WebRTC W | WebRTC A | public-webrtc | RESO | WONT | Media loopback testing | 2013-01-29 | 2014-10-28 |
20818 | WebRTC W | WebRTC A | public-webrtc | RESO | MOVE | TCP media unspecified | 2013-01-29 | 2014-04-29 |
20819 | WebRTC W | WebRTC A | public-webrtc | RESO | LATE | no priority API | 2013-01-29 | 2014-10-31 |
20820 | WebRTC W | WebRTC A | public-webrtc | RESO | FIXE | congestion control of data and audio/video | 2013-01-29 | 2014-10-28 |
21082 | WebRTC W | WebRTC A | public-webrtc | RESO | WORK | remoteDescription and localDescription should return NULL if the PeerConnection has been closed | 2013-02-22 | 2013-02-26 |
21086 | WebRTC W | WebRTC A | public-webrtc | RESO | LATE | getLocalStreams and getRemoteStreams should return empty sequence after Peerconnection::close | 2013-02-22 | 2014-12-15 |
21877 | WebRTC W | WebRTC A | martin.thomson | RESO | LATE | API is unable to handle inbound streams prior to arrival of answer | 2013-04-30 | 2014-12-15 |
21878 | WebRTC W | WebRTC A | public-webrtc | RESO | FIXE | Unable to learn of unknown inbound media | 2013-04-30 | 2014-10-31 |
21879 | WebRTC W | WebRTC A | public-webrtc | RESO | LATE | Unable to access certificate information in the API | 2013-04-30 | 2014-12-15 |
21880 | WebRTC W | WebRTC A | martin.thomson | RESO | LATE | Certificate management is underspecified | 2013-04-30 | 2014-12-15 |
21950 | WebRTC W | WebRTC A | public-webrtc | RESO | FIXE | Add success/error callbacks to addIceCandidate (and possibly other API calls) | 2013-05-07 | 2014-10-29 |
22347 | WebRTC W | WebRTC A | public-webrtc | RESO | FIXE | RTCIceServer should have multiple URLs | 2013-06-13 | 2014-10-29 |
22428 | WebRTC W | WebRTC A | public-webrtc | RESO | WONT | Please unify the WebSocket and RTCDataChannel "readyState". enum RTCDataChannelState { "connecting", "open", "closing", "closed" }; VS const unsigned short CONNECTING = 0 [...] | 2013-06-23 | 2013-12-06 |
22441 | WebRTC W | WebRTC A | martin.thomson | RESO | LATE | Bug in section 8.1.2 Requesting Assertions | 2013-06-25 | 2014-12-15 |
22442 | WebRTC W | WebRTC A | martin.thomson | RESO | LATE | Bug in section 8.1.3 Verifying Assertions | 2013-06-25 | 2014-12-15 |
23572 | WebRTC W | WebRTC A | public-webrtc | RESO | FIXE | Documented format with which to specify ICE servers does not match implementation and contains typos | 2013-10-20 | 2014-10-29 |
23832 | WebRTC W | WebRTC A | public-webrtc | RESO | WONT | Requiring that negotiated channels be created on the receiver before any data can be received is problematic for some use cases | 2013-11-14 | 2014-10-31 |
23919 | WebRTC W | WebRTC A | adam.bergkvist | RESO | LATE | DataChannel.onerror callback needs an error argument specified. | 2013-11-26 | 2014-12-15 |
23920 | WebRTC W | WebRTC A | juberti | RESO | LATE | TURN authentication failures should be surfaced as some event | 2013-11-26 | 2014-12-15 |
24061 | WebRTC W | WebRTC A | public-webrtc | RESO | FIXE | Need to update TURN / STUN URI references | 2013-12-11 | 2014-10-29 |
24875 | WebRTC W | WebRTC A | public-webrtc | RESO | FIXE | Examples in the WebRTC spec are not updated As per the modified API. | 2014-03-01 | 2014-06-04 |
25102 | WebRTC W | WebRTC A | adam.bergkvist | RESO | LATE | RTCDataChannel::send() steps are not proper. | 2014-03-20 | 2014-12-15 |
25152 | WebRTC W | WebRTC A | public-webrtc | RESO | FIXE | createObjectURL used in examples is no longer supported by Media Capture and Streams. | 2014-03-26 | 2014-06-09 |
25155 | WebRTC W | WebRTC A | adam.bergkvist | RESO | FIXE | maxRetransmitTime is not the name of the SCTP concept it points to | 2014-03-26 | 2014-06-10 |
25189 | WebRTC W | WebRTC A | public-webrtc | RESO | FIXE | Mandatory errorCallback is missing in examples for getStats. | 2014-03-28 | 2014-06-09 |
25257 | WebRTC W | WebRTC A | public-webrtc | RESO | WONT | Explanation missing for ICE connection state transition from connected to disconnected state. | 2014-04-04 | 2014-06-09 |
25440 | WebRTC W | WebRTC A | dom | RESO | LATE | MediaStreamTrack.readyState has no muted attribute | 2014-04-24 | 2014-12-15 |
25497 | WebRTC W | WebRTC A | juberti | RESO | LATE | RTCRtpSender / Receiver objects need to be added to the specification | 2014-04-29 | 2014-12-15 |
25513 | WebRTC W | WebRTC A | adam.bergkvist | RESO | LATE | WebRTC spec should explicitly specify all causes of a PeerConnection-sourced track being muted | 2014-04-30 | 2014-12-15 |
25531 | WebRTC W | WebRTC A | public-webrtc | RESO | FIXE | Validation for requestIdentity attribute is missing. | 2014-05-02 | 2014-10-30 |
25533 | WebRTC W | WebRTC A | ekr | RESO | LATE | WebRTC spec should explicitly specify the state transition for cancelled offers. | 2014-05-02 | 2014-12-15 |
25544 | WebRTC W | WebRTC A | public-webrtc | RESO | WONT | Options attribute of createOffer / createAnswer should be validated before processing. | 2014-05-05 | 2014-10-31 |
25545 | WebRTC W | WebRTC A | public-webrtc | RESO | INVA | Initialization of of RTCConfiguration while invoking RTCPeerConnection.getConfiguration should be updated. | 2014-05-05 | 2014-10-31 |
25576 | WebRTC W | WebRTC A | adam.bergkvist | RESO | WONT | steps for createDTMFSender() are missing. | 2014-05-06 | 2014-11-05 |
25579 | WebRTC W | WebRTC A | public-webrtc | RESO | WONT | State transitions are missing in RTCPeerConnections state transition diagram. | 2014-05-06 | 2014-10-31 |
25596 | WebRTC W | WebRTC A | juberti | RESO | LATE | updateIce should be called setConfiguration | 2014-05-08 | 2014-12-15 |
25724 | WebRTC W | WebRTC A | adam.bergkvist | RESO | FIXE | Allow garbage collection of closed PeerConnections | 2014-05-15 | 2016-01-19 |
25806 | WebRTC W | WebRTC A | pthatcher | RESO | LATE | ice pool size | 2014-05-19 | 2014-12-15 |
25807 | WebRTC W | WebRTC A | public-webrtc | RESO | NEED | Avoid sdpMangling by modifying codec preferences through API. | 2014-05-19 | 2014-05-19 |
25808 | WebRTC W | WebRTC A | juberti | RESO | LATE | add new acces for the active remote/local SDP | 2014-05-19 | 2014-12-15 |
25811 | WebRTC W | WebRTC A | public-webrtc | RESO | WORK | Change extensible enum to dom strings | 2014-05-19 | 2014-10-31 |
25828 | WebRTC W | WebRTC A | martin.thomson | RESO | LATE | Need to add pc.canTrickle) | 2014-05-20 | 2014-12-15 |
25833 | WebRTC W | WebRTC A | martin.thomson | RESO | LATE | change the definition of "enqueue a task" as EKR slides May 20 | 2014-05-20 | 2014-12-15 |
25834 | WebRTC W | WebRTC A | martin.thomson | RESO | LATE | close is synchronous & idempotent | 2014-05-20 | 2014-12-15 |
25835 | WebRTC W | WebRTC A | martin.thomson | RESO | LATE | when closing, all outstanding actions are cancelled and their callbacks are fired with a "cancelled" error | 2014-05-20 | 2014-12-15 |
25836 | WebRTC W | WebRTC A | public-webrtc | RESO | LATE | add note about addtrack being async | 2014-05-20 | 2014-12-15 |
25837 | WebRTC W | WebRTC A | public-webrtc | RESO | FIXE | throw error when setting invalid parameters to DTMF sender | 2014-05-20 | 2014-06-18 |
25840 | WebRTC W | WebRTC A | public-webrtc | RESO | FIXE | Creating DataChannel with same label. | 2014-05-20 | 2014-06-09 |
25841 | WebRTC W | WebRTC A | public-webrtc | RESO | WONT | DataChannel and DTMF methods should check for peerConnection closed state | 2014-05-20 | 2014-05-21 |
25855 | WebRTC W | WebRTC A | public-webrtc | RESO | FIXE | Clarification about conformance requirements phrased as algorithms | 2014-05-21 | 2014-06-10 |
25856 | WebRTC W | WebRTC A | martin.thomson | RESO | LATE | Add way to find out if a MST is isolated or becomes isolated | 2014-05-21 | 2014-12-15 |
25859 | WebRTC W | WebRTC A | martin.thomson | RESO | LATE | Streams that become isolated generate errors on PC | 2014-05-21 | 2014-12-15 |
25892 | WebRTC W | WebRTC A | public-webrtc | RESO | FIXE | SignalingStateChange event should be fired only if there is a change in signaling state. | 2014-05-27 | 2014-06-09 |
25893 | WebRTC W | WebRTC A | public-webrtc | RESO | WONT | Offer Answer options should supported sendOnly and inactive media states. | 2014-05-27 | 2014-10-31 |
25957 | WebRTC W | WebRTC A | juberti | RESO | LATE | PeerConnection should have an onerror event handler | 2014-06-03 | 2014-12-15 |
25975 | WebRTC W | WebRTC A | public-webrtc | RESO | WONT | When can the value of DTMFSender.canInsertDTMF change? | 2014-06-04 | 2014-10-31 |
25976 | WebRTC W | WebRTC A | public-webrtc | RESO | FIXE | DTMFSender.insertDTMF steps should validate the values of duration and interToneGap. | 2014-06-04 | 2014-06-09 |
25977 | WebRTC W | WebRTC A | public-webrtc | RESO | WONT | DTMFSender should have an onerror event handler | 2014-06-04 | 2014-06-11 |
26027 | WebRTC W | WebRTC A | martin.thomson | RESO | LATE | addIceCandidate should not be callable when PeerConnection is closed | 2014-06-10 | 2014-12-15 |
26279 | WebRTC W | WebRTC A | juberti | RESO | LATE | Options attribute is required for createAnswer | 2014-07-08 | 2014-12-15 |
26364 | WebRTC W | WebRTC A | public-webrtc | RESO | DUPL | Add "rollback" to RTCSdpType | 2014-07-17 | 2014-10-31 |
26620 | WebRTC W | WebRTC A | vsingh.ietf | RESO | MOVE | getStats should be allowed on a closed PeerConnection. | 2014-08-20 | 2014-10-31 |
26644 | WebRTC W | WebRTC A | martin.thomson | RESO | LATE | MID in Candidate event attributes | 2014-08-22 | 2014-12-15 |
27211 | WebRTC W | WebRTC A | pthatcher | RESO | LATE | Add BundlePolicy to RTCConfiguration | 2014-10-31 | 2014-12-15 |
27213 | WebRTC W | WebRTC A | pthatcher | RESO | LATE | DTMFSender should hang off RTCRTPSender, not MediaStreamTrack | 2014-10-31 | 2014-12-15 |
27214 | WebRTC W | WebRTC A | martin.thomson | RESO | LATE | ICE gathering state change should surface an event | 2014-10-31 | 2014-12-15 |
27224 | WebRTC W | WebRTC A | juberti | RESO | LATE | Transport objects should be added to the specification | 2014-11-03 | 2014-12-15 |
27225 | WebRTC W | WebRTC A | juberti | RESO | LATE | Encoding Parameters need to be added to RTPSender object | 2014-11-03 | 2014-12-15 |
27226 | WebRTC W | WebRTC A | juberti | RESO | LATE | Codec capabilities need to be accessible on an RTPSender | 2014-11-03 | 2014-12-15 |
27227 | WebRTC W | WebRTC A | public-webrtc | RESO | LATE | New feature: Track swapping | 2014-11-03 | 2014-12-15 |
27704 | WebRTC W | WebRTC A | dom | NEW | --- | Dictionary RTCStats should use type DOMHighResTimeStamp instead of DOMHiResTimeStamp | 2014-12-26 | 2014-12-26 |